Signal processing system and method for calibrating channel signals supplied from an array of sensors having different operating characteristics

ABSTRACT

In a signal processing system, a set of channel signals from an array of sensors of different operating characteristics are processed in calibration circuitry that calculates individual average values of the channel signals and calculates an average of the individual average values of channel signals as a reference value. Reciprocal calculators calculate reciprocal values of the individual average values of the channel signals. Scaling circuitry scales the reciprocal values by the reference value to produce a set of amplitude calibration signals and scales the channel signals by the calibration signals respectively. As a result, the channel signals are normalized by their own average values and scaled by the reference value to produce a set of calibrated channel signals.

CROSS-REFERENCE TO RELATED PATENT APPLICATIONS

This application is a divisional of application Ser. No. 11/514,201filed Sep. 1, 2006, now pending, which claims the benefit of priorityfrom the prior Japanese Patent Application No. 2005-255158, filed Sep.2, 2005, the entire contents of which are incorporated herein byreference. This application claims only subject matter disclosed in theparent application and therefore presents no new matter.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to a signal processing system forcalibrating multiple channel signals from an array of sensors, such asmicrophones and antennas, having different operating characteristics.

2. Description of the Related Art

It is well known to use an array of sensors, such as microphones andradio antennas equally spaced at predetermined intervals and oriented ina specified direction for cancelling interference signals by forming abeam in the direction of arrival of a target signal. When microphonesare used as the array sensors, the Griffiths-Jim beamformer is the basictechnique also known as a generalized sidelobe canceller. In theGriffiths-Jim beamformer as described in a literature “MicrophoneArrays,” Springer, 2001, pages 87 to 109, signals from the microphonearray are combined in a fixed beamformer to enhance a desired signal andattenuate interference signals. The array beam is formed by the fixedbeamfonner by linearly summing the microphone signals. If allmicrophones have the same operating characteristic, the summationresults in an output that is M times the magnitude of each microphonesignal (where M is the number of microphones). Hence the signals thatarrive perpendicularly to the array surface (i.e., broadside signals)can be constructively combined. Since signals arriving in otherdirections have timing (phase) differences from the broadside signals,they interact with each other in a destructive manner. As a result, ifthe signals arriving perpendicularly to the array surface are the targetsignal, the target signal is enhanced, and the microphone array producesa directivity in a direction normal to its surface.

The microphone signals are also applied to a blocking matrix where thesesignals are combined to obtain multiple interference references. Theenhanced target signal is delayed for an interval corresponding to thetime taken by the blocking matrix to perform the matrix calculation. Thedelayed beamformer output and the interference references are combinedin a multi-channel canceller. In the multi-channel canceller theinterference references are used as interference replicas forsubtraction from the enhanced target signal to produce an enhancedtarget signal.

However, if the operating characteristics of the microphones are notequal to each other due to their variation, the microphone signalspartially interact with each other in a destructive way. This results inthe array producing a degraded directivity in the direction normal toits surface. A similar problem occurs in the blocking matrix. In thiscase, it is the target signal that finds a leakage path to the outputsof the blocking matrix. This results in the multi-channel cancellerperforming partial cancellation of the target signal and causes adistortion in its output signal.

The element imperfection problem, caused by the above-mentionedvariation, is addressed by a calibration technique described in IEEETransactions on Signal Processing, Vol. 42, No. 10, pages 2871-2875,October 1994. According to this technique, a blocking matrix is designedbased on optimal eigenvector constraints by using a broadband signal anda fixed beamformer corresponding to the blocking matrix is thendesigned. However, the use of broadband signal requires that individualcalibration is necessary for each microphone array in advance of itsmanufacture. This is disadvantageous for quantity production.

Another calibration technique, as described in IEEE Transactions onAntennas and Propagations, Vol. 34, No. 8, pages 996-1012, August 1986,introduces noise at an appropriate level to each microphone signal.However, this prior art requires precision setting of the added noiselevel. Data such as signal-to-noise ratios, interference-to-signalratios and the level of noise at each microphone must be additionallycalculated on a real-time basis. The amount of computations issubstantial and the additive noise is a potential source of poor soundquality. Further calibration techniques are disclosed in a number ofpatent publications. Japanese Patent Publication 2004-343700 discloses atechnique that uses a calibration speaker and a signal processing systemand Japanese Patent 3337671 teaches the use of a calibration microphone.Japanese Patent Publication 2002-502193 employs multiple adaptivefilters for respective microphones. However, these prior art techniquesrequire separate devices that present an increase in hardware. Accordingto a further technique disclosed in Japanese Patent Publication2004-502367, the output of a single microphone is used as a referencelevel. However, this reference level must be supplied from an externalsource.

SUMMARY OF THE INVENTION

It is therefore an object of the present invention to provide acalibration system and method for an array of sensors having differentoperating characteristics, which uses only input signals to performcalibration without advance calibration and additional computations andhardware.

According to a first aspect of the present invention, there is provideda signal processing system for processing a plurality of channel signalssupplied from an array of sensors of different operatingcharacteristics, comprising calibration circuitry that determines areference value from the channel signals, equally splits the channelsignals into first copies of the channel signals and second copies ofthe channel signals, and produces calibrated channel signals that areequal to the first copies normalized respectively by average values ofthe second copies and scaled by the reference value.

According to a second aspect, the present invention provides a signalprocessing system for processing a plurality of channel signals suppliedfrom an array of sensors of different operating characteristics,comprising a plurality of analysis filter banks that respectivelyreceive the channel signals from the sensors, wherein each analysisfilter bank decomposes the received channel signal into a plurality ofsubband-channel signals of different frequencies, a plurality ofmulti-channel equalizers corresponding in number to the subband-channelsignals decomposed by each analysis filter bank, wherein eachmulti-channel equalizer receives a plurality of subband-channel signalsof same frequency from all of the analysis filter banks and produces aplurality of calibrated subband-channel signals, and a plurality ofsynthesis filter banks that respectively receive the calibratedsubband-channel signals of different frequencies from all of themulti-channel equalizers, wherein each synthesis filter bank composesthe calibrated subband-channel signals into a calibrated channel signal.Each of the multi-channel equalizers comprises calibration circuitrythat determines a reference value from the received subband-channelsignals, equally splits the subband-channel signals into first copies ofsubband-channel signals and second copies of subband-channel signals,and produces calibrated subband-channel signals that are equal to thefirst copies normalized respectively by average values of the secondcopies and scaled by said reference value.

According to a third aspect, the present invention provides a signalprocessing system processing a plurality of channel signals suppliedfrom an array of sensors of different operating characteristics,comprising: a plurality of transform circuits that respectively receivethe channel signals from the sensors, wherein each transform circuittransforms the received channel signal to a frequency-domain signalhaving amplitudes and phases of a plurality of different frequencycomponents of the received channel signal, and a plurality ofmulti-channel equalizers corresponding in number to different frequencycomponents of the frequency-domain signal transformed by each transformcircuit, wherein each multi-channel equalizer receives a plurality ofsame frequency components of the frequency-domain signals from all ofthe transform circuits and produces a plurality of calibrated samefrequency components of a frequency-domain signals. Each of themulti-channel equalizers comprises calibration circuitry that determinesa reference value from the received same frequency components, equallysplits the same frequency components into first copies of the samefrequency components and second copies of the same frequency components,and produces calibrated same frequency components that are equal to thefirst copies normalized respectively by average values of the secondcopies and scaled by said reference value.

According to a fourth aspect, the present invention provides a method ofprocessing a plurality of channel signals supplied from an array ofsensors of different operating characteristics, comprising the step ofcalibrating the channel signals by determining a reference value fromthe channel signals, equally splitting the channel signals into firstcopies of the channel signals and second copies of the channel signals,and producing calibrated channel signals that are equal to the firstcopies normalized respectively by average values of the second copiesand scaled by said reference value.

According to a fifth aspect, the present invention provides a method ofprocessing a plurality of channel signals supplied from an array ofsensors of different operating characteristics, comprising (a)decomposing each of the channel signals into a plurality ofsubband-channel signals of different frequencies, (b) calibrating samefrequency subband-channel signals of each of the channel signals bydetermining a reference value from the same frequency subband-channelsignals, equally splitting the subband-channel signals into first copiesof the same frequency subband-channel signals and second copies of thesame frequency subband-channel signals, and producing calibratedsubband-channel signals that are equal to the first copies normalizedrespectively by average values of the second copies and scaled by saidreference value, and (c) composing calibrated subband-channel signals ofdifferent frequencies into a plurality of calibrated channel signals.

According to a sixth aspect, the present invention provides a method ofprocessing a plurality of channel signals supplied from an array ofsensors of different operating characteristics, comprising (a)transforming each of the channel signals from the sensors to afrequency-domain signal having amplitudes and phases of a plurality ofdifferent frequency components, and (b) calibrating same frequencycomponents of each of the frequency-domain signals by determining areference value from the same frequency components, equally splittingthe same frequency components into first copies of the same frequencycomponents and second copies of the same frequency components, andproducing calibrated same frequency components that are equal to thefirst copies normalized respectively by average values of the secondcopies and scaled by said reference value.

BRIEF DESCRIPTION OF THE DRAWINGS

The present invention will be described in detail with reference to thefollowing drawings, in which:

FIG. 1 is a block diagram of a signal processing system of a firstembodiment of the present invention for an array of microphones;

FIG. 2 is a block diagram of the multi-channel equalizer of FIG. 1;

FIG. 3 is a block diagram of the gain calculator of FIG. 2;

FIG. 4 is a block diagram of a signal processing system according to asecond embodiment of the present invention;

FIG. 5 is a block diagram of a first form of the subband multi-channelequalizer of FIG. 4;

FIG. 6 is a block diagram of a second form of the frequency-domainmulti-channel equalizer of FIG. 4;

FIG. 7 is a block diagram of a third form of the frequency-domainmulti-channel equalizer of FIG. 4;

FIG. 8 is a block diagram of each multi-channel phase equalizer of FIG.7;

FIG. 9 is a block diagram of a fourth form of the frequency-domainmulti-channel equalizer of FIG. 4;

FIG. 10 is a block diagram of each multi-channel phase equalizer of FIG.9;

FIG. 11 is a block diagram of a signal processing system according to athird embodiment of the present invention; and

FIG. 12 is a block diagram of a signal processing system according to afourth embodiment of the present invention.

DETAILED DESCRIPTION

In FIG. 1, there is shown a signal processing system according to afirst embodiment of the present invention for an array of sensors thatare equi-spaced at predetermined intervals along the surface of thearray. For the purpose of disclosure, an array of microphones 100 ₀˜100_(M−1) are used as sensors. A fixed beamformer 200, a blocking matrix300 and a multi-channel canceller 500 are provided in the same manner asthe corresponding elements of the Griffiths-Jim Beamformer describedpreviously.

According to the present invention, a multi-channel equalizer 700 isprovided to process microphone (or channel) signals from the array.Multi-channel equalizer 700 receives the input channel signalsx₀˜x_(M−1) from microphones 100 ₀˜100 _(M−1) and produces calibratedmicrophone signals y₀˜y_(M−1) which are supplied to the fixed beamformer200 and the blocking matrix 300.

In the beamfonner, the calibrated microphone signals are linearly summedtogether to produce an enhanced target signal. The enhanced targetsignal is in a delay circuit 400 and supplied to the multi-channelcanceller 500.

As described below the microphone signals are processed in the equalizer700 in such a manner that they behave as if they were produced bycalibrated microphones. Therefore, the summation in the beamformer 200results in an output that is exactly M times the magnitude of eachmicrophone signal. Hence the signals that arrive perpendicularly to thesurface of the array are constructively combined and signals arriving inother directions are destructively combined. As a result, if the surfaceof the array is normal to the arriving direction of a target signal, thesignals arriving in that direction are constructively combined and anenhanced target signal is produced at the output of the fixed beamformer200.

Blocking matrix 300 produces multiple interference references from thecalibrated microphone signals. In the multi-channel canceller, theinterference references are used as interference replicas forsubtraction from the enhanced target signal to produce an enhancedtarget signal.

FIG. 2 illustrates the details of the multi-channel equalizer 700.Equalizer 700 comprises a gain calculator 710 and a plurality ofmultipliers 711 ₀˜711 _(M−1). Microphone signals x₀˜x_(M−1) from thearray are equally split into first copies of the channel signals andsecond copies of the channel signals. The first copies of the channelsignals are respectively supplied to the multipliers 711 and the secondcopies of the channel signals are supplied to the gain calculator 710.Gain calculator 710 produces a plurality of gains g₀˜g_(M−1) which aresupplied respectively to the multipliers 711 ₀˜711 _(M−1). Thecalibrated channel signals y₀˜y_(M−1) are produced in the multipliers711 ₀˜711 _(M−1) by respectively scaling the first channel signalsx₀˜x_(M−1) by the gains g₀˜g_(M−1).

As shown in FIG. 3, the gain calculator 710 is comprised of a pluralityof average calculators 712 ₀˜712 _(M−1) connected respectively to themicrophones 100 ₀˜100 _(M−1) to receive the second channel signals andproduce individual average values x ₀˜ x _(M−1) of the microphonesignals x₀˜x_(M−1). One averaging method is based on asliding-window-based smoothing calculation known in the signalprocessing art as finite impulse response filters. If an L-tap (sample)sliding widow is used, the average power of each channel signal isobtained as follows:

$\begin{matrix}{{\overset{\_}{x}}_{k}^{2} = {\frac{1}{L}{\sum\limits_{j = {k - L + 1}}^{k}x_{j}^{2}}}} & (1)\end{matrix}$

where k is an index representing the current time.

Another averaging method involves the use of a first order leakyintegrator, which yields the following average value:

x ² _(k) =β· x ² _(k−1)+(1−β)·x ² _(k)   (2)

where β is a constant and satisfies the relation 0<β<1. Note that if themicrophone signals contain speech, it is preferable to set the integer Lequal to a sample value that corresponds to a period of 2 to 3 secondsand set the constant β so that it corresponds to L.

The average values x ₀˜ x _(M−1) are supplied to reciprocal calculators713 ₀˜713 _(M−1). Reciprocal calculators 713 ₀˜713 _(M−1) calculate thereciprocals of the average values x ₀˜ x _(M−1) and then calculate theirsquare roots as follows and supply them to multipliers 714 ₀˜714 _(M−1).

$\begin{matrix}{\sqrt{\frac{1}{{\overset{\_}{x}}_{0}^{2}}},\sqrt{\frac{1}{{\overset{\_}{x}}_{1}^{2}}},\ldots \mspace{14mu},\sqrt{\frac{1}{{\overset{\_}{x}}_{M - 1}^{2}}}} & (3)\end{matrix}$

The average values x ₀˜ x _(M−1) of the microphone signals are alsosupplied to a reference calculator 715, which produces a reference(average) value s₀ of the individual average values of the channelsignals by calculating the following Equation:

$\begin{matrix}{s_{0} = \frac{\sqrt{\sum\limits_{j = 0}^{M - 1}{\overset{\_}{x}}_{j}^{2}}}{\sqrt{M}}} & (4)\end{matrix}$

The reference value s₀ is an average value of the individual averagepower values of all channels calculated by the average calculators 712.This average value s₀ is used in the multipliers 714 ₀˜714 _(M−1) tomultiply the outputs of the reciprocal calculators 713 ₀˜713 _(M−1) toproduce gain values g_(i) (where i=0, 1, . . . , M−1) as follows:

$\begin{matrix}{g_{i} = {\frac{1}{\sqrt{M}}\sqrt{\frac{\sum\limits_{j = 0}^{M - 1}{\overset{\_}{x}}_{j}^{2}}{{\overset{\_}{x}}_{i}^{2}}}}} & (5)\end{matrix}$

By multiplying the reciprocals of the individual average values ofchannel (microphone) signals by the reference value s₀, the gain(calibration) signal g_(i) has the effect of normalizing the channelsignals with their own average values and of calibrating the channelsignals with the reference signal. Equation (5) indicates that the gainvalue for calibrating each channel signal can be easily obtained bysimple calculations of average and square root of the channel signal.

Therefore, the multi-channel equalizer 700 produces calibrated channelsignals that are equal to the first copies of the channel signalsnormalized with the average values of the second copies and scaled bythe reference signal jointly by the multipliers 711 ₀˜711 ¹⁻¹ andmultipliers 714 ₀˜714 _(M−1). As a result, the calibration can be madewith a small amount of computations.

In FIG. 2, the calibration (gain) signals g₀˜g_(M−1) are used inmultipliers 711 ₀˜711 _(M−1) to scale the channel signals x₀˜x_(M−1),respectively. As a result, the channel signals are calibrated in such away that they behave as if they were generated by an array of sensors ofequal operating characteristics.

FIG. 4 is a block diagram of a second embodiment of the presentinvention in which a subband multi-channel equalizer 800 is used insteadof the multi-channel equalizer 700 of FIG. 1. In FIG. 4, partscorresponding to those of FIG. 1 bear the same numerals and thedescription thereof is omitted for simplicity.

As shown in detail in FIG. 5, the subband multi-channel equalizer 800-1of FIG. 4 according to a first form of the present invention comprises Manalysis filter banks 810 ₀˜810 _(M−1) which are respectively connectedto the microphones 100 ₀˜100 _(M−1), to receive the channel signalsx₀˜x_(M−1). Each analysis filter bank 810, decomposes the frequencyspectrum of the corresponding channel signal x, into N subband-channelsignals x_(i,j) (where i=0, 1, . . . , M−1, j=0, 1, . . . , N−1)representing the different frequency components, or subbands of thechannel signal x_(i).

A plurality of multi-channel equalizers 700 ₀˜700 _(N−1) are providedcorresponding in number to the N subband-channels decomposed by eachanalysis filter bank. Each of these multi-channel equalizers 700 ₀˜700_(N−1) is of identical configuration to that shown in FIG. 1 andprovided with M input terminals corresponding in number to the Manalysis filter banks 810.

All subband-channel signals of the analysis filter banks 810 ₀˜810_(M−1) are supplied to the multi-channel equalizers 700 ₀˜700 _(N−1) insuch a manner that the multi-channel equalizer 700 _(i) receives Msubband-channel signals x_(i,0), . . . , x,_(i,j), . . . , x_(i,M−1) ofthe same frequency band “i” from all analysis filter banks. Eachmulti-channel equalizer 700 equalizes its M subband-channel signals toproduce M calibrated subband-channel signals in a manner identical tothat described in connection with the previous embodiment.

A plurality of synthesis filter banks 820 ₀˜820 _(M−1), are provided,each having N input terminals. Synthesis filter bank 820 receives Mcalibrated subband-channel signals of the different frequency componentsfrom all multi-channel equalizers 700 ₀˜700 _(N−1) to produce acalibrated channel signal y_(j).

In the analysis filter banks, the spectrum of each channel signal may bedivided into subband-channel of uniform subbands or nonuniform subbands.In the latter case, if the lower frequency range of the channel spectrumis divided into narrower bandwidths and the higher frequency range isdivided into broader bandwidths, time-domain resolution can be low inthe lower frequency range and high in the higher frequency range. Asdescribed in “Multirate Systems and Filter Banks, Prentice-Hall, 1993,pages 45-60, 188-393, 478-479, the channel spectrum may be dividedaccording to octave division in which the bandwidth of eachsubband-channel is one-half of the bandwidth of its higher-frequencyadjacent subband-channel. The channel spectrum may also be divided usingthe critical band which is based on the human hearing characteristics.

FIG. 6 illustrates the subband multi-channel equalizer 800-2 of FIG. 4according to a second form of the present invention, in which partscorresponding to those in FIG. 5 bear the same reference numerals.Instead of using analysis filter banks of FIG. 5, this embodiment usesfrequency domain converters, or transform circuits 811 ₀˜811 _(M−1).Each transform circuit 811 _(i) (where i=0, 1, . . . , M−1) transformsthe output signal of the associated microphone to a frequency-domainsignal which shows the amplitude and phase of different frequencycomponents (i.e., sub-channels) x_(j,0)˜x_(j,N−1) of its input channelsignal x_(j). The amplitude information of the different frequencycomponents are separated from the corresponding phase information.

Transform circuit 811, supplies each one of its amplitude signalsx_(i,0)˜x_(j,N−1) to a corresponding input terminal “j” of each of the Nmulti-channel equalizers 700 ₀˜700 _(N−1) so that the equalizer 700 _(j)(where j=0, 1, N−1) receives amplitude signals |x_(i,M−1)| of the samefrequency component x, from all transform circuits 811. Multi-channelequalizer 700 _(j) performs calibration (equalization) on the M inputamplitude signals in the same way as described previously, and producesa set of M calibrated amplitude output signals that appear at outputterminals “0” to “M−1.”

Multi-channel equalizer 700 _(j) supplies each one of its calibratedamplitude signals to a corresponding amplitude input terminal “j” ofeach of N-input time-domain converters, or inverse transform circuits821 ₀˜821 _(M−1) so that the inverse transform circuit 821 _(i) receivescalibrated amplitude signals |x_(i,0)|, . . . , |x_(i,j)|, . . . ,|x_(i,N−1)| of all frequency components from the multi-channelequalizers 700 ₀˜700 _(N−1).

The phase information separated from the transform circuit 810 _(j) isdirectly supplied to the inverse transform circuit 821 _(j), to which Mcalibrated amplitude signals of different frequency components are alsosupplied from all multi-channel equalizers 700 ₀˜700 _(N−1). Using thephase signals, the inverse transform circuit 821 _(j) combines thereceived amplitude signals to synthesize a frequency domain channelsignal and performs inverse transform on the synthesized frequencydomain signal to produce a calibrated time-domain channel signal y_(j).

In a practical aspect, the transform circuits 811 ₀˜811 _(M−1) organizea plurality of input samples into a number of blocks and performtransform on each of the blocks. In a similar manner, the inversetransform circuits perform inverse transform on the same number of inputsamples. Examples of the transform are Fourier transform, cosinetransform and Karhunen-Loeve transform. Details of these transforms aredescribed in “Digital Coding of Waveforms, Principles and Applicationsto Speech and Video,” Prentice-Hall, 1990, pages 510-563.

In a further aspect, a windowing technique may be used in the process offrequency domain conversion. In this case, the transform circuits 811₀˜811 _(M−1) multiply each block of input samples by a window function,such as Hamming, Hanning (or Han), Kaiser, or Blackman window, andperform the transform on the windowed input samples. Details of thesewindow functions are described in literatures “Multirate Systems andFilter Banks,” Prentice-Hall, 1993, pages 45-60, 188-393, 478-479, and“Digital Signal Processing,” Prentice-Hall, 1975, pages 239-250.

In a still further aspect, the organized blocks of input samples may bepartially overlapped with adjacent blocks. For example, if 30 percent ofthe block length is overlapped, the last 30 percent of a block is usedas the first 30 percent of the next block. Corresponding to the wayblocks of samples are partially overlapped in the transform circuits811, the blocks are partially overlapped in the inverse transformcircuits 821. Further information for overlapped transform is describedin a literature “Digital Coding of Waveforms, Principles andApplications to Speech and Video,” Prentice-Hall, 1990, pages 510-563.

FIG. 7 illustrates each frequency-domain multi-channel equalizer 800-3of FIG. 4 according to a third form of the present invention, in whichparts corresponding to those in FIG. 6 bear the same reference numerals.This embodiment differs from FIG. 6 in that a plurality of multi-channelphase equalizers 701 ₀˜701 _(M−1), are additionally providedcorresponding to the multi-channel amplitude equalizers 700 ₀˜700 _(M−1)for equalizing the phase information of the corresponding amplitude ofeach frequency component.

Similar to the manner the multi-channel amplitude equalizers 700 ₀˜700_(N−1) are connected between the amplitude output terminals of transformcircuits 811 ₀˜811 _(M−1) and the amplitude input terminals of inversetransform circuits 821 ₀˜821 _(M−1), the multi-channel phase equalizers701 ₀˜701 _(M−1) are connected between the phase output terminals oftransform circuits 811 ₀˜811 _(M−1) and the phase input terminals ofinverse transform circuits 821 ₀˜821 _(M−1). Therefore, the transformcircuit 811 _(i) (where i=0, 1, . . . , M−1) supplies the phaseinformation of frequency components x_(i,0)˜x_(i,N−1) to a correspondinginput terminal “j” of each of the N multi-channel phase equalizers 701₀˜701 _(M−1) so that the phase equalizer 701 _(j)(where j=0, 1, . . . ,N−1) receives phase signals <x_(i,0), . . . <x_(i,j), . . . <x_(i,N−1)of the same frequency component x_(i) from all transform circuits 811.

Multi-channel phase equalizer 701 _(j) supplies each one of itscalibrated phase signals to a corresponding phase input terminal “j” ofeach of N-input inverse transform circuits 821 ₀˜821 _(M−1) so that theinverse transform circuit 821, receives calibrated phase signals<y_(i,j), . . . , <y_(i,N−1) of all frequency components from themulti-channel phase equalizers 701 ₀˜701 _(M−1). Using the calibratedphase signals, the inverse transform circuit 821 _(j) combines thereceived amplitude signals to synthesize a frequency-domain channelsignal and performs inverse transform on the frequency domain signal toproduce a calibrated time-domain channel signal y_(i).

As shown in FIG. 8, each of the multi-channel phase equalizers 701 ₀˜701_(M−1) comprises a plurality of adders 717 ₀˜717 _(M−1) and a phasecorrection calculator 716 which includes a plurality of average phaseangle calculators 718 ₀˜718 _(M−1). In the multi-channel phase equalizer701 _(j), the phase information signals <x₀˜<x_(M−1) of the samefrequency component “j” are supplied from all transform circuits 811₀˜811 _(M−1) respectively to both adders 717 ₀˜717 _(M−1) and averagephase angle calculators 718 ₀˜718 _(M−1). Average phase anglecalculators 718 ₀˜718 _(M−1) produce average phase angle values < x ₀, .. . < x _(M−1), which are further averaged by a reference calculator 719to produce a reference (average) phase angle value <G0, which is givenas follows:

$\begin{matrix}{{\angle \; G_{0}} = \frac{\sum\limits_{j = 0}^{M - 1}{\angle {\overset{\_}{x}}_{j}}}{M}} & (6)\end{matrix}$

The average values < x ₀, . . . < x _(M−1) are respectively subtractedin subtractors 720 ₀˜720 _(M−1), from the reference value <GO to producea set of phase correction values as given below:

<g _(i) =<G ₀ −< x _(i) (where i=1, 0, . . . , M−1)   (7)

The phase correction values <g₀˜<g_(M−1) are respectively combined inthe adders 717 ₀˜717 _(M−1) with their input phase information signals<x₀˜<x_(M−1) to produce corrected phase information signals <y₀˜<y_(M−1)as given by Equation (8):

<y _(i) =<x _(i) +<g _(i)   (8)

As a result, when the phase correction values <g₀˜<g_(M−1) are used inthe adders 717 ₀˜717 _(M−1) to shift the phase information signals<x₀˜<x_(M−1), the phase information signals <y₀˜<y_(M−1) are equallyaligned (calibrated) in time with each other.

If the phase correction value <g_(i) is negative, the minimum (i.e.,maximum of the negative values) of a correction value <ĝ_(i) (which isgiven below) is first determined and the correction value <g_(i) iscorrected so that the minimum value of the correction value =ĝ_(i) isreduced to zero. Eventually, the correction value <ĝ_(i) is expressed byEquation (9).

<ĝ _(i) =<g _(i)−min{<g _(i)}  (9)

A modified frequency-domain multi-channel equalizer 800-4 is shown inFIG. 9 which differs from the embodiment of FIG. 7 in that amplitudesignals |x₀|˜|x_(M−1)| are also used in calibrating the phaseinformation. Multi-channel phase equalizers 702 ₀˜702 _(N−1) receive theamplitude signals |x₀|˜|x_(M−1)| from the transform circuits 811 ₀˜811_(M−1) , along with the phase information signals <x₀˜<x_(M−1).

As shown in FIG. 10, each multi-channel phase equalizer 702 _(j) issimilar to that shown in FIG. 8 with the exception that the amplitudesignals |x₀|˜|x_(M−1)| are supplied to a phase correction calculator721, while the phase signals are only supplied to the adders 717 ₀˜717_(M−1).

Phase correction calculator 721 differs from the phase correctioncalculator 716 of FIG. 8 in that it additionally includes a relativedelay calculator 722 to determine the delay time differences δ₀˜δ_(M−1)between the input amplitude signals and feeds the relative delay timevalues to the average calculators 718 ₀˜718 _(M−1) to calculate theiraverage value δ _(i). The outputs of average calculators 718 aresupplied to the reference calculator 719 and the subtractors 720 ₀˜720_(M−1). More specifically, in the relative delay calculator 721 therelative delay time of a given amplitude signal is determined by firstselecting one of the input amplitude signals as a reference amplitudesignal. Correlation between the given amplitude signal and the referencesignal is then calculated. This correlation calculation is repeated bysuccessively shifting the timing point of correlation until thecalculated correlation increases to a maximum, whereupon the relativedelay time of the given amplitude signal is determined.

Reference calculator 719 calculates the average value δ₀ of the averagedrelative delay time difference values as follows:

$\begin{matrix}{\delta_{0} = \frac{\sum\limits_{j = 0}^{M - 1}{\overset{\_}{\delta}}_{j}}{M}} & (10)\end{matrix}$

The average delay time difference value δ₀ is supplied to thesubtractors 720 ₀˜720 _(M−1) to produce the phase correction values asgiven below:

<g _(i)=2πf(δ₀−δ_(i)) (where i=0, 1, . . . , M−1)   (11)

If −δ_(i)+δ₀ is negative, the minimum of −δ_(i)+δ₀ is first determinedand the value (−δ_(i)+δ₀) is corrected so that the minimum valueequivalently corresponds to zero. Eventually, the correction value<ĝ_(i) is expressed by Equation (9).

<ĝ _(i)=2πf{δ ₀−δ_(i)−min(δ₀−δ_(i))}  (12)

In a further aspect of the present invention, a frequency-domainmulti-channel equalizer 900 is shown in FIG. 11 as a modification of thefrequency-domain multi-channel equalizers 800-3 and 800-4. In thismodification, a transform domain adaptive filter can be used instead ofthe inverse transform circuits 821 ₀˜821 _(M−1).

Each of the transform circuitry 901 ₀˜901 _(M−1) generates a set of Nfrequency-domain signals x₀˜x_(N−1) and the multi-channel amplitudeequalizers 700 ₀˜700 _(N−1) are provided in number corresponding to thenumber of amplitude signals generated by each of the transform circuits901 ₀˜901 _(M−1). Likewise, the multi-channel phase equalizers 701 ₀˜701_(N−1) or 702 ₀˜702 _(N−1) are provided in number corresponding to thenumber of phase information signals generated by each of the transformcircuits 901 ₀˜901 _(M−1). The blocking matrix 300 described previouslyreceives N calibrated amplitude signals |y₀|˜|y_(M−1)| and N calibratedphase signals <y₀˜<y_(M−1) to perform transform-domain adaptivefiltering.

Note that the transform domain adaptive filters suitable for use in thepresent invention are described in a literature “Adaptive Filters,” JohnWiley & Sons, 1998, pages 201-292.

FIG. 12 shows a further modified embodiment of the present invention. Inthis modification, direction-of-arrival estimation circuitry 201 isprovided instead of the fixed beamformer 200, blocking matrix 300, delayelement 400 and multi-channel canceller 500. The DOA estimationcircuitry 201 performs estimation of the arrival direction of incomingchannel signals that are calibrated by the multi-channel equalizer 700or 800 described in the previous embodiments. Based on the phasedifferences (relative delays) between the calibrated channel signals todetermine in which direction the incoming signals are arriving accordingto the algorithm known in the art. Examples of the known algorithm arefound in literatures “IEICE Transactions on Fundamentals,” Vol. 87-A,No. 3, pages 559-566, March 2004, and “IEICE Transactions onFundamentals,” Vol. 88-A, No. 3, pages 633-641, March 2005.

Multi-channel equalizers 700 and 800 described above can beadvantageously used in combination with the fixed beamformer 200,blocking matrix 300 and multi-channel canceller 500. By adaptivelyprocessing the calibrated channel signals supplied from themulti-channel equalizer, the fixed beamformer 200 forms a beam onsignals arriving on the array of sensors 100 in a predetermineddirection, while the blocking matrix 300 forms a null on signals thatarrive in other direction. Multi-channel canceller 500 that adaptivelyprocesses the channel signals by using the beam and the nullrespectively formed by the beamformer 200 and blocking matrix 300.Alternatively, a generalized sidelobe canceller is implemented with acombination of the fixed beamformer 200, the null forming circuitry, orblocking matrix 300 and the multi-channel canceller 500.

1. A signal processing system for processing a plurality of channel signals supplied from an array of sensors of different operating characteristics, comprising: a plurality of analysis filter banks that respectively receive said channel signals from said sensors, wherein each analysis filter bank decomposes the received channel signal into a plurality of subband-channel signals of different frequencies; a plurality of multi-channel equalizers corresponding in number to said subband-channel signals decomposed by each analysis filter bank, wherein each multi-channel equalizer receives a plurality of subband-channel signals of same frequency from all of said analysis filter banks and produces a plurality of calibrated subband-channel signals; and a plurality of synthesis filter banks that respectively receive said calibrated subband-channel signals of different frequencies from all of said multi-channel equalizers, wherein each synthesis filter bank composes the calibrated subband-channel signals into a calibrated channel signal, wherein each of said multi-channel equalizers comprises calibration circuitry that determines a reference value from the received subband-channel signals, equally splits said subband-channel signals into first copies of said subband-channel signals and second copies of said subband-channel signals, and produces calibrated subband-channel signals that are equal to said first copies normalized respectively by average values of said second copies and scaled by said reference value.
 2. The signal processing system of claim 1, wherein said calibration circuitry comprises: reference calculation circuitry that calculates individual average values of said subband-channel signals and determines said reference value from said individual average values; reciprocal calculation circuitry that calculates reciprocal values of said individual average values; and scaling circuitry that scales said reciprocal values by said reference value to produce a plurality of amplitude calibration signals and scales said subband-channel signals by said calibration signals respectively.
 3. The signal processing system of claim 1, wherein said reference value is an average value of said individual average values.
 4. The signal processing system of claim 3, wherein said individual average values are individual average power values of said subband-channel signals and said average value of said individual average is average power of said individual average power values.
 5. The signal processing system of claim 1, further comprising beamforming circuitry that forms a beam on signals arriving on said sensor array in a predetermined direction by using the calibrated channel signals from said synthesis filter banks.
 6. The signal processing system of claim 5, further comprising null forming circuitry that forms a null on signals arriving on said sensor array in a predetermined direction by using the calibrated channel signals from said synthesis filter banks and a canceller that adaptively processes said calibrated channel signals by using said beam and said null.
 7. The signal processing system of claim 5, wherein said beamforming circuitry adaptively performs the forming of said beam.
 8. The signal processing system of claim 6, wherein said null forming circuitry adaptively performs the forming of said null.
 9. The signal processing system of claim 6, wherein a combination of said beamforming circuitry, said null forming circuitry and said canceller is a generalized sidelobe canceller.
 10. The signal processing system of claim 1, further comprising estimation circuitry for estimating the direction of arrival of signals on said array of sensors by using said calibrated channel signals.
 11. A signal processing system for processing a plurality of channel signals supplied from an array of sensors of different operating characteristics, comprising: a plurality of transform circuits that respectively receive said channel signals from said sensors, wherein each transform circuit transforms the received channel signal to a frequency-domain signal having amplitudes and phases of a plurality of different frequency components of the received channel signal; and a plurality of multi-channel equalizers corresponding in number to different frequency components of said frequency-domain signal transformed by each transform circuit, wherein each multi-channel equalizer receives a plurality of same frequency components of said frequency-domain signals from all of said transform circuits and produces a plurality of calibrated same frequency components of a frequency-domain signals. wherein each of said multi-channel equalizers comprises calibration circuitry that determines a reference value from said received same frequency components, equally splits said same frequency components into first copies of said same frequency components and second copies of said same frequency components, and produces calibrated same frequency components that are equal to said first copies normalized respectively by average values of said second copies and scaled by said reference value.
 12. The signal processing system of claim 11, further comprising a plurality of inverse transform circuits, wherein each inverse transform circuit receives calibrated different frequency components of said frequency-domain signals from all of said multi-channel equalizers and transforms the calibrated different frequency components into a plurality of calibrated channel signals.
 13. The signal processing system of claim 11, wherein said calibration circuitry comprises: reference calculation circuitry that calculates individual average values of said same frequency components and determines said reference value from said individual average values; reciprocal calculation circuitry that calculates reciprocal values of said individual average values; and scaling circuitry that scales said reciprocal values by said reference value to produce a plurality of amplitude calibration signals and scales said same frequency components of said frequency-domain signals by said calibration signals respectively.
 14. The signal processing system of claim 13, wherein said reference value is an average value of said individual average values.
 15. The signal processing system of claim 13, wherein said individual average values are individual average power values of said same frequency components and said average value of said individual average values is average power of said individual average power values.
 16. The signal processing system of claim 11, further comprising a plurality of multi-channel phase equalizers corresponding in number to said different frequency components of each frequency-domain signal transformed by each of said transform circuits wherein each multi-channel phase equalizer receives a plurality of phase information signals of same frequency components of said frequency-domain signals from all of said transform circuits and produces a plurality of calibrated phase information signals of same frequency components, wherein each of said multi-channel phase equalizers comprises: phase correction circuitry that calculates average phase angle values of phase information signals of said same frequency components, and determines a reference phase angle value from said average phase angle values; and phase shifting circuitry that phase shifts the average phase angle values of said phase information signals with said reference phase angle value to produce a plurality of phase calibration signals and phase shifts said phase information signals of same frequency components with said phase calibration signals, respectively.
 17. The signal processing system of claim 16, further comprising a plurality of inverse transform circuits, wherein each inverse transform circuit receives calibrated different frequency components of said frequency-domain signals from all of said multi-channel equalizers and phase-shifted different frequency components from all of said multi-channel phase equalizers and transforms the received different frequency components by using the received phase-shifted different frequency components into a plurality of calibrated channel signals.
 18. The signal processing system of claim 16, wherein said reference phase angle value is an average value of said phase angle values.
 19. The signal processing system of claim 11, further comprising a plurality of multi-channel phase equalizers corresponding in number to said different frequency components transformed by each transform circuit, wherein each multi-channel phase equalizer receives a plurality of phase information signals and amplitude signals of said same frequency components from all of said transform circuits and produces a plurality of calibrated phase information signals of same frequency components, wherein each of said multi-channel phase equalizers comprises: phase correction circuitry that determines relative delay time difference values between said amplitude signals of said same frequency components and determines a reference delay time difference value from said relative delay time difference values; and phase shifting circuitry that phase shifts the relative delay time difference values with said reference delay time difference value to produce a plurality of phase calibration signals and phase shifts said phase information signals of same frequency components with said phase calibration signals, respectively.
 20. The signal processing system of claim 19, further comprising a plurality of inverse transform circuits, wherein each inverse transform circuit receives calibrated different frequency components of said frequency-domain signals from all of said multi-channel equalizers and phase-shifted different frequency components from all of said multi-channel phase equalizers and transforms the received different frequency components by using the received phase-shifted different frequency components into a plurality of calibrated channel signals.
 21. The signal processing system of claim 19, wherein said reference delay time difference value is an average value of said relative delay time difference values.
 22. The signal processing system of claim 17, further comprising beamforming circuitry that forms a beam on signals arriving on said sensor array in a predetermined direction by using the calibrated channel signals from said inverse transform circuits.
 23. The signal processing system of claim 22, further comprising null forming circuitry that forms a null on signals arriving on said sensor array in a predetermined direction by using the calibrated channel signals from said synthesis filter banks and a canceller that adaptively processes said calibrated channel signals by using said beam and said null.
 24. The signal processing system of claim 22, wherein said beamforming circuitry adaptively performs the forming of said beam.
 25. The signal processing system of claim 23, wherein said null forming circuitry adaptively performs the forming of said null.
 26. The signal processing system of claim 23, wherein a combination of said beamforming circuitry, said null forming circuitry and said canceller is a generalized sidelobe canceller.
 27. The signal processing system of claim 17, further comprising estimation circuitry for estimating the direction of arrival of signals on said array of sensors by using said calibrated channel signals.
 28. A method of processing a plurality of channel signals supplied from an array of sensors of different operating characteristics, comprising: a) decomposing each of said channel signals into a plurality of subband-channel signals of different frequencies; b) calibrating same frequency subband-channel signals of each of said channel signals by determining a reference value from said same frequency subband-channel signals, equally splitting said same frequency subband-channel signals into first copies of said same frequency subband-channel signals and second copies of said same frequency subband-channel signals, and producing calibrated same frequency components that are equal to said first copies normalized respectively by average values of said second copies and scaled by said reference value; and c) composing said calibrated subband-channel signals of different frequencies into a plurality of calibrated channel signals.
 29. The method of claim 28, wherein step (b) comprises the steps of: calculating individual average values of said same frequency subband-channel signals and deter mining a reference value from said individual average values; calculating reciprocal values of said individual average values; scaling said reciprocal values by said reference value to produce a plurality of amplitude calibration signals and scaling said subband-channel signals by said calibration signals respectively to produce a plurality of calibrated subband-channel signals.
 30. The method of claim 29, wherein said reference value is an average value of said individual average values of subband-channel signals.
 31. The method of claim 30, wherein said individual average values are individual average power values of said subband-channel signals and said average value of said individual average values is average power of said individual average power values.
 32. The method of claim 28, further comprising the step of fol. ling a beam on signals arriving on said sensor array in a predetermined direction by using the calibrated channel signals.
 33. The method of claim 32, further comprising the steps of forming a null on signals arriving on said sensor array in a predetermined direction by using the calibrated channel signals and adaptively processing said calibrated channel signals by using said beam and said null.
 34. The method of claim 32, wherein the step of forming beam adaptively performs the forming of said beam.
 35. The method of claim 33, wherein said null forming step adaptively performs the forming of said null.
 36. The method of claim 33, wherein a combination of said beamforming step, said null forming step and said adaptively processing step is performed by using a generalized sidelobe canceller.
 37. The method of claim 28, further comprising estimating the direction of arrival of signals on said array of sensors by using said calibrated channel signals.
 38. A method of processing a plurality of channel signals supplied from an array of sensors of different operating characteristics, comprising: a) transforming each of said channel signals from said sensors to a frequency-domain signal having amplitudes and phases of a plurality of different frequency components; and b) calibrating same frequency components of each of said frequency-domain signals by determining a reference value from said same frequency components, equally splitting the same frequency components into first copies of the same frequency components and second copies of the same frequency components, and producing calibrated same frequency components that are equal to said first copies normalized respectively by average values of said second copies and scaled by said reference value.
 39. The method of claim 38, wherein step (b) comprises: calculating individual average values of said same frequency components of each frequency-domain signal and determining a reference value from said individual average values; calculating reciprocal values of said individual average values; and scaling said reciprocal values by said reference value to produce a plurality of amplitude calibration signals and scaling said same frequency components by said amplitude calibration signals respectively to produce a plurality of calibrated same frequency components of said frequency-domain signals.
 40. The method of claim 38, further comprising inversely transforming said calibrated different frequency components.
 41. The method of claim 39, wherein said reference value is an average value of said individual average values.
 42. The method of claim 40, wherein said individual average values are individual average power values of said same frequency components and said average value of said individual average values is average power of said individual average power values.
 43. The method of claim 38, further comprising: calculating average phase angle values of phase information signals of said same frequency components; determining a reference phase angle value from said average phase angle values; phase shifting the average phase angle values of said phase information signals with said reference phase angle value to produce a plurality of phase calibration signals; and phase shifting said phase information signals with said phase calibration signals, respectively.
 44. The method of claim 43, wherein said reference phase angle value is an average value of said phase angle values.
 45. The method of claim 43, further comprising inversely transforming said calibrated different frequency components using said phase shifted phase information signals.
 46. The method of claim 45, further comprising: determining relative delay time difference values between amplitude signals of said same frequency components; determining a reference delay time difference value from said relative delay time difference values; phase shifting the relative delay time difference values with said reference delay time difference value to produce a plurality of phase calibration signals; and phase shifting said phase information signals with said phase calibration signals, respectively.
 47. The method of claim 46, wherein said reference delay time difference value is an average value of said relative delay time difference values.
 48. The method of claim 46, further comprising inversely transforming said calibrated different frequency components using said phase shifted phase information signals.
 49. The method of claim 40, further comprising the step of forming a beam on signals arriving on said sensor array in a predetermined direction by using the calibrated different frequency components.
 50. The method of claim 49, further comprising the steps of forming a null on signals arriving on said sensor array in a predetermined direction by using the calibrated channel signals and adaptively processing said calibrated different frequency components by using said beam and said null.
 51. The method of claim 49, wherein the step of forming beam adaptively performs the forming of said beam.
 52. The method of claim 50, wherein said null forming step adaptively performs the forming of said null.
 53. The method of claim 50, wherein a combination of said beamforming step, said null foaming step and said adaptively processing step is performed by using a generalized sidelobe canceller.
 54. The method of claim 40, further comprising estimating the direction of arrival of signals on said array of sensors by using said calibrated channel signals. 